add conformer online server, test=doc

paddlespeech/cli/asr/infer.py

@@ -91,6 +91,20 @@ pretrained_models = {
         'https://deepspeech.bj.bcebos.com/zh_lm/zh_giga.no_cna_cmn.prune01244.klm',
         'lm_md5':
         '29e02312deb2e59b3c8686c7966d4fe3'
+    },
+        "conformer2online_aishell-zh-16k": {
+        'url':
+        'https://paddlespeech.bj.bcebos.com/s2t/aishell/asr1/asr1_chunk_conformer_aishell_ckpt_0.1.2.model.tar.gz',
+        'md5':
+        '4814e52e0fc2fd48899373f95c84b0c9',
+        'cfg_path':
+        'config.yaml',
+        'ckpt_path':
+        'exp/deepspeech2_online/checkpoints/avg_30',
+        'lm_url':
+        'https://deepspeech.bj.bcebos.com/zh_lm/zh_giga.no_cna_cmn.prune01244.klm',
+        'lm_md5':
+        '29e02312deb2e59b3c8686c7966d4fe3'
     },
     "deepspeech2offline_librispeech-en-16k": {
         'url':
@@ -115,6 +129,8 @@ model_alias = {
     "paddlespeech.s2t.models.ds2_online:DeepSpeech2ModelOnline",
     "conformer":
     "paddlespeech.s2t.models.u2:U2Model",
+    "conformer2online":
+    "paddlespeech.s2t.models.u2:U2Model",
     "transformer":
     "paddlespeech.s2t.models.u2:U2Model",
     "wenetspeech":
@@ -219,6 +235,7 @@ class ASRExecutor(BaseExecutor):
         """
         Init model and other resources from a specific path.
         """
+        logger.info("start to init the model")
         if hasattr(self, 'model'):
             logger.info('Model had been initialized.')
             return
@@ -233,14 +250,15 @@ class ASRExecutor(BaseExecutor):
             self.ckpt_path = os.path.join(
                 res_path, pretrained_models[tag]['ckpt_path'] + ".pdparams")
             logger.info(res_path)
-            logger.info(self.cfg_path)
-            logger.info(self.ckpt_path)
+
         else:
             self.cfg_path = os.path.abspath(cfg_path)
             self.ckpt_path = os.path.abspath(ckpt_path + ".pdparams")
             self.res_path = os.path.dirname(
                 os.path.dirname(os.path.abspath(self.cfg_path)))
-
+        logger.info(self.cfg_path)
+        logger.info(self.ckpt_path)
+        
         #Init body.
         self.config = CfgNode(new_allowed=True)
         self.config.merge_from_file(self.cfg_path)
@@ -269,7 +287,6 @@ class ASRExecutor(BaseExecutor):
                     vocab=self.config.vocab_filepath,
                     spm_model_prefix=self.config.spm_model_prefix)
                 self.config.decode.decoding_method = decode_method
-
             else:
                 raise Exception("wrong type")
         model_name = model_type[:model_type.rindex(
@@ -347,12 +364,14 @@ class ASRExecutor(BaseExecutor):
         else:
             raise Exception("wrong type")
 
+        logger.info("audio feat process success")
+
     @paddle.no_grad()
     def infer(self, model_type: str):
         """
         Model inference and result stored in self.output.
         """
-
+        logger.info("start to infer the model to get the output")
         cfg = self.config.decode
         audio = self._inputs["audio"]
         audio_len = self._inputs["audio_len"]
@@ -369,17 +388,22 @@ class ASRExecutor(BaseExecutor):
             self._outputs["result"] = result_transcripts[0]
 
         elif "conformer" in model_type or "transformer" in model_type:
-            result_transcripts = self.model.decode(
-                audio,
-                audio_len,
-                text_feature=self.text_feature,
-                decoding_method=cfg.decoding_method,
-                beam_size=cfg.beam_size,
-                ctc_weight=cfg.ctc_weight,
-                decoding_chunk_size=cfg.decoding_chunk_size,
-                num_decoding_left_chunks=cfg.num_decoding_left_chunks,
-                simulate_streaming=cfg.simulate_streaming)
-            self._outputs["result"] = result_transcripts[0][0]
+            logger.info(f"we will use the transformer like model : {model_type}")
+            try:
+                result_transcripts = self.model.decode(
+                    audio,
+                    audio_len,
+                    text_feature=self.text_feature,
+                    decoding_method=cfg.decoding_method,
+                    beam_size=cfg.beam_size,
+                    ctc_weight=cfg.ctc_weight,
+                    decoding_chunk_size=cfg.decoding_chunk_size,
+                    num_decoding_left_chunks=cfg.num_decoding_left_chunks,
+                    simulate_streaming=cfg.simulate_streaming)
+                self._outputs["result"] = result_transcripts[0][0]
+            except Exception as e:
+                logger.exception(e)
+
         else:
             raise Exception("invalid model name")
 

paddlespeech/s2t/models/u2/u2.py

@@ -213,12 +213,14 @@ class U2BaseModel(ASRInterface, nn.Layer):
                 num_decoding_left_chunks=num_decoding_left_chunks
             )  # (B, maxlen, encoder_dim)
         else:
+            print("offline decode from the asr")
             encoder_out, encoder_mask = self.encoder(
                 speech,
                 speech_lengths,
                 decoding_chunk_size=decoding_chunk_size,
                 num_decoding_left_chunks=num_decoding_left_chunks
             )  # (B, maxlen, encoder_dim)
+            print("offline decode success")
         return encoder_out, encoder_mask
 
     def recognize(
@@ -706,13 +708,15 @@ class U2BaseModel(ASRInterface, nn.Layer):
             List[List[int]]: transcripts.
         """
         batch_size = feats.shape[0]
+        print("start to decode the audio feat")
         if decoding_method in ['ctc_prefix_beam_search',
                                'attention_rescoring'] and batch_size > 1:
-            logger.fatal(
+            logger.error(
                 f'decoding mode {decoding_method} must be running with batch_size == 1'
             )
+            logger.error(f"current batch_size is {batch_size}")
             sys.exit(1)
-
+        print(f"use the {decoding_method} to decode the audio feat")
         if decoding_method == 'attention':
             hyps = self.recognize(
                 feats,

paddlespeech/s2t/modules/ctc.py

@@ -180,7 +180,8 @@ class CTCDecoder(CTCDecoderBase):
         # init once
         if self._ext_scorer is not None:
             return
-
+        
+        from paddlespeech.s2t.decoders.ctcdecoder import Scorer  # noqa: F401
         if language_model_path != '':
             logger.info("begin to initialize the external scorer "
                         "for decoding")

paddlespeech/s2t/modules/encoder.py

@@ -317,6 +317,8 @@ class BaseEncoder(nn.Layer):
         outputs = []
         offset = 0
         # Feed forward overlap input step by step
+        print(f"context: {context}")
+        print(f"stride: {stride}")
         for cur in range(0, num_frames - context + 1, stride):
             end = min(cur + decoding_window, num_frames)
             chunk_xs = xs[:, cur:end, :]

paddlespeech/server/conf/ws_application.yaml

@@ -4,7 +4,7 @@
 #                             SERVER SETTING                                    #
 #################################################################################
 host: 0.0.0.0
-port: 8091
+port: 8096
 
 # The task format in the engin_list is: <speech task>_<engine type>
 # task choices = ['asr_online', 'tts_online']
@@ -18,10 +18,44 @@ engine_list: ['asr_online']
 #                                ENGINE CONFIG                                  #
 #################################################################################
 
+# ################################### ASR #########################################
+# ################### speech task: asr; engine_type: online #######################
+# asr_online:
+#     model_type: 'deepspeech2online_aishell'
+#     am_model: # the pdmodel file of am static model [optional]
+#     am_params:  # the pdiparams file of am static model [optional]
+#     lang: 'zh'
+#     sample_rate: 16000
+#     cfg_path: 
+#     decode_method: 
+#     force_yes: True
+
+#     am_predictor_conf:
+#         device:  # set 'gpu:id' or 'cpu'
+#         switch_ir_optim: True
+#         glog_info: False  # True -> print glog
+#         summary: True  # False -> do not show predictor config
+
+#     chunk_buffer_conf:
+#         frame_duration_ms: 80
+#         shift_ms: 40
+#         sample_rate: 16000
+#         sample_width: 2
+
+#     vad_conf:
+#         aggressiveness: 2
+#         sample_rate: 16000
+#         frame_duration_ms: 20
+#         sample_width: 2
+#         padding_ms: 200
+#         padding_ratio: 0.9
+
+
+
 ################################### ASR #########################################
 ################### speech task: asr; engine_type: online #######################
 asr_online:
-    model_type: 'deepspeech2online_aishell'
+    model_type: 'conformer2online_aishell'
     am_model: # the pdmodel file of am static model [optional]
     am_params:  # the pdiparams file of am static model [optional]
     lang: 'zh'
@@ -37,15 +71,15 @@ asr_online:
         summary: True  # False -> do not show predictor config
 
     chunk_buffer_conf:
-        frame_duration_ms: 80
+        frame_duration_ms: 85
         shift_ms: 40
         sample_rate: 16000
         sample_width: 2
 
-    vad_conf:
-        aggressiveness: 2
-        sample_rate: 16000
-        frame_duration_ms: 20
-        sample_width: 2
-        padding_ms: 200
-        padding_ratio: 0.9
+    # vad_conf:
+    #     aggressiveness: 2
+    #     sample_rate: 16000
+    #     frame_duration_ms: 20
+    #     sample_width: 2
+    #     padding_ms: 200
+    #     padding_ratio: 0.9

paddlespeech/server/engine/asr/online/asr_engine.py

@@ -20,11 +20,15 @@ from numpy import float32
 from yacs.config import CfgNode
 
 from paddlespeech.cli.asr.infer import ASRExecutor
+from paddlespeech.cli.asr.infer import model_alias
+from paddlespeech.cli.asr.infer import pretrained_models
 from paddlespeech.cli.log import logger
 from paddlespeech.cli.utils import MODEL_HOME
 from paddlespeech.s2t.frontend.featurizer.text_featurizer import TextFeaturizer
 from paddlespeech.s2t.frontend.speech import SpeechSegment
 from paddlespeech.s2t.modules.ctc import CTCDecoder
+from paddlespeech.s2t.transform.transformation import Transformation
+from paddlespeech.s2t.utils.dynamic_import import dynamic_import
 from paddlespeech.s2t.utils.utility import UpdateConfig
 from paddlespeech.server.engine.base_engine import BaseEngine
 from paddlespeech.server.utils.audio_process import pcm2float
@@ -51,6 +55,24 @@ pretrained_models = {
         'lm_md5':
         '29e02312deb2e59b3c8686c7966d4fe3'
     },
+    "conformer2online_aishell-zh-16k": {
+        'url':
+        'https://paddlespeech.bj.bcebos.com/s2t/aishell/asr0/asr1_chunk_conformer_aishell_ckpt_0.1.2.model.tar.gz',
+        'md5':
+        '4814e52e0fc2fd48899373f95c84b0c9',
+        'cfg_path':
+        'exp/chunk_conformer//conf/config.yaml',
+        'ckpt_path':
+        'exp/chunk_conformer/checkpoints/avg_30/',
+        'model':
+        'exp/chunk_conformer/checkpoints/avg_30.pdparams',
+        'params':
+        'exp/chunk_conformer/checkpoints/avg_30.pdparams',
+        'lm_url':
+        'https://deepspeech.bj.bcebos.com/zh_lm/zh_giga.no_cna_cmn.prune01244.klm',
+        'lm_md5':
+        '29e02312deb2e59b3c8686c7966d4fe3'
+    },
 }
 
 
@@ -71,15 +93,17 @@ class ASRServerExecutor(ASRExecutor):
         """
         Init model and other resources from a specific path.
         """
-
+        self.model_type = model_type
+        self.sample_rate = sample_rate
         if cfg_path is None or am_model is None or am_params is None:
             sample_rate_str = '16k' if sample_rate == 16000 else '8k'
             tag = model_type + '-' + lang + '-' + sample_rate_str
             logger.info(f"Load the pretrained model, tag = {tag}")
             res_path = self._get_pretrained_path(tag)  # wenetspeech_zh
             self.res_path = res_path
-            self.cfg_path = os.path.join(res_path,
-                                         pretrained_models[tag]['cfg_path'])
+            self.cfg_path = "/home/users/xiongxinlei/task/paddlespeech-develop/PaddleSpeech/paddlespeech/server/tests/asr/online/conf/config.yaml"
+            # self.cfg_path = os.path.join(res_path,
+            #                              pretrained_models[tag]['cfg_path'])
 
             self.am_model = os.path.join(res_path,
                                          pretrained_models[tag]['model'])
@@ -119,49 +143,67 @@ class ASRServerExecutor(ASRExecutor):
                     lm_url,
                     os.path.dirname(self.config.decode.lang_model_path), lm_md5)
             elif "conformer" in model_type or "transformer" in model_type or "wenetspeech" in model_type:
-                # 开发 conformer 的流式模型
                 logger.info("start to create the stream conformer asr engine")
-                # 复用cli里面的代码
-
+                if self.config.spm_model_prefix:
+                    self.config.spm_model_prefix = os.path.join(
+                        self.res_path, self.config.spm_model_prefix)
+                self.config.vocab_filepath = os.path.join(
+                    self.res_path, self.config.vocab_filepath)
+                self.text_feature = TextFeaturizer(
+                    unit_type=self.config.unit_type,
+                    vocab=self.config.vocab_filepath,
+                    spm_model_prefix=self.config.spm_model_prefix)
+                # update the decoding method
+                if decode_method:
+                    self.config.decode.decoding_method = decode_method
             else:
                 raise Exception("wrong type")
-
-        # AM predictor
-        logger.info("ASR engine start to init the am predictor")
-        self.am_predictor_conf = am_predictor_conf
-        self.am_predictor = init_predictor(
-            model_file=self.am_model,
-            params_file=self.am_params,
-            predictor_conf=self.am_predictor_conf)
-
-        # decoder
-        logger.info("ASR engine start to create the ctc decoder instance")
-        self.decoder = CTCDecoder(
-            odim=self.config.output_dim,  # <blank> is in  vocab
-            enc_n_units=self.config.rnn_layer_size * 2,
-            blank_id=self.config.blank_id,
-            dropout_rate=0.0,
-            reduction=True,  # sum
-            batch_average=True,  # sum / batch_size
-            grad_norm_type=self.config.get('ctc_grad_norm_type', None))
-
-        # init decoder
-        logger.info("ASR engine start to init the ctc decoder")
-        cfg = self.config.decode
-        decode_batch_size = 1  # for online
-        self.decoder.init_decoder(
-            decode_batch_size, self.text_feature.vocab_list,
-            cfg.decoding_method, cfg.lang_model_path, cfg.alpha, cfg.beta,
-            cfg.beam_size, cfg.cutoff_prob, cfg.cutoff_top_n,
-            cfg.num_proc_bsearch)
-
-        # init state box
-        self.chunk_state_h_box = np.zeros(
-            (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
-            dtype=float32)
-        self.chunk_state_c_box = np.zeros(
-            (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
-            dtype=float32)
+        if "deepspeech2online" in model_type or "deepspeech2offline" in model_type:
+            # AM predictor
+            logger.info("ASR engine start to init the am predictor")
+            self.am_predictor_conf = am_predictor_conf
+            self.am_predictor = init_predictor(
+                model_file=self.am_model,
+                params_file=self.am_params,
+                predictor_conf=self.am_predictor_conf)
+
+            # decoder
+            logger.info("ASR engine start to create the ctc decoder instance")
+            self.decoder = CTCDecoder(
+                odim=self.config.output_dim,  # <blank> is in  vocab
+                enc_n_units=self.config.rnn_layer_size * 2,
+                blank_id=self.config.blank_id,
+                dropout_rate=0.0,
+                reduction=True,  # sum
+                batch_average=True,  # sum / batch_size
+                grad_norm_type=self.config.get('ctc_grad_norm_type', None))
+
+            # init decoder
+            logger.info("ASR engine start to init the ctc decoder")
+            cfg = self.config.decode
+            decode_batch_size = 1  # for online
+            self.decoder.init_decoder(
+                decode_batch_size, self.text_feature.vocab_list,
+                cfg.decoding_method, cfg.lang_model_path, cfg.alpha, cfg.beta,
+                cfg.beam_size, cfg.cutoff_prob, cfg.cutoff_top_n,
+                cfg.num_proc_bsearch)
+
+            # init state box
+            self.chunk_state_h_box = np.zeros(
+                (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
+                dtype=float32)
+            self.chunk_state_c_box = np.zeros(
+                (self.config.num_rnn_layers, 1, self.config.rnn_layer_size),
+                dtype=float32)
+        elif "conformer" in model_type or "transformer" in model_type or "wenetspeech" in model_type:
+            model_name = model_type[:model_type.rindex(
+                '_')]  # model_type: {model_name}_{dataset}
+            logger.info(f"model name: {model_name}")
+            model_class = dynamic_import(model_name, model_alias)
+            model_conf = self.config
+            model = model_class.from_config(model_conf)
+            self.model = model
+            logger.info("create the transformer like model success")
 
     def reset_decoder_and_chunk(self):
         """reset decoder and chunk state for an new audio
@@ -186,6 +228,7 @@ class ASRServerExecutor(ASRExecutor):
         Returns:
             [type]: [description]
         """
+        logger.info("start to decoce chunk by chunk")
         if "deepspeech2online" in model_type:
             input_names = self.am_predictor.get_input_names()
             audio_handle = self.am_predictor.get_input_handle(input_names[0])
@@ -224,10 +267,29 @@ class ASRServerExecutor(ASRExecutor):
 
             self.decoder.next(output_chunk_probs, output_chunk_lens)
             trans_best, trans_beam = self.decoder.decode()
+            logger.info(f"decode one one best result: {trans_best[0]}")
             return trans_best[0]
 
         elif "conformer" in model_type or "transformer" in model_type:
-            raise Exception("invalid model name")
+            try:
+                logger.info(
+                    f"we will use the transformer like model : {self.model_type}"
+                )
+                cfg = self.config.decode
+                result_transcripts = self.model.decode(
+                    x_chunk,
+                    x_chunk_lens,
+                    text_feature=self.text_feature,
+                    decoding_method=cfg.decoding_method,
+                    beam_size=cfg.beam_size,
+                    ctc_weight=cfg.ctc_weight,
+                    decoding_chunk_size=cfg.decoding_chunk_size,
+                    num_decoding_left_chunks=cfg.num_decoding_left_chunks,
+                    simulate_streaming=cfg.simulate_streaming)
+
+                return result_transcripts[0][0]
+            except Exception as e:
+                logger.exception(e)
         else:
             raise Exception("invalid model name")
 
@@ -244,32 +306,55 @@ class ASRServerExecutor(ASRExecutor):
         """
         # pcm16 -> pcm 32
         samples = pcm2float(samples)
-
-        # read audio
-        speech_segment = SpeechSegment.from_pcm(
-            samples, sample_rate, transcript=" ")
-        # audio augment
-        self.collate_fn_test.augmentation.transform_audio(speech_segment)
-
-        # extract speech feature
-        spectrum, transcript_part = self.collate_fn_test._speech_featurizer.featurize(
-            speech_segment, self.collate_fn_test.keep_transcription_text)
-        # CMVN spectrum
-        if self.collate_fn_test._normalizer:
-            spectrum = self.collate_fn_test._normalizer.apply(spectrum)
-
-        # spectrum augment
-        audio = self.collate_fn_test.augmentation.transform_feature(spectrum)
-
-        audio_len = audio.shape[0]
-        audio = paddle.to_tensor(audio, dtype='float32')
-        # audio_len = paddle.to_tensor(audio_len)
-        audio = paddle.unsqueeze(audio, axis=0)
-
-        x_chunk = audio.numpy()
-        x_chunk_lens = np.array([audio_len])
-
-        return x_chunk, x_chunk_lens
+        if "deepspeech2online" in self.model_type:
+            # read audio
+            speech_segment = SpeechSegment.from_pcm(
+                samples, sample_rate, transcript=" ")
+            # audio augment
+            self.collate_fn_test.augmentation.transform_audio(speech_segment)
+
+            # extract speech feature
+            spectrum, transcript_part = self.collate_fn_test._speech_featurizer.featurize(
+                speech_segment, self.collate_fn_test.keep_transcription_text)
+            # CMVN spectrum
+            if self.collate_fn_test._normalizer:
+                spectrum = self.collate_fn_test._normalizer.apply(spectrum)
+
+            # spectrum augment
+            audio = self.collate_fn_test.augmentation.transform_feature(
+                spectrum)
+
+            audio_len = audio.shape[0]
+            audio = paddle.to_tensor(audio, dtype='float32')
+            # audio_len = paddle.to_tensor(audio_len)
+            audio = paddle.unsqueeze(audio, axis=0)
+
+            x_chunk = audio.numpy()
+            x_chunk_lens = np.array([audio_len])
+
+            return x_chunk, x_chunk_lens
+        elif "conformer2online" in self.model_type:
+
+            if sample_rate != self.sample_rate:
+                logger.info(f"audio sample rate {sample_rate} is not match," \
+                            "the model sample_rate is {self.sample_rate}")
+            logger.info(f"ASR Engine use the {self.model_type} to process")
+            logger.info("Create the preprocess instance")
+            preprocess_conf = self.config.preprocess_config
+            preprocess_args = {"train": False}
+            preprocessing = Transformation(preprocess_conf)
+
+            logger.info("Read the audio file")
+            logger.info(f"audio shape: {samples.shape}")
+            # fbank
+            x_chunk = preprocessing(samples, **preprocess_args)
+            x_chunk_lens = paddle.to_tensor(x_chunk.shape[0])
+            x_chunk = paddle.to_tensor(
+                x_chunk, dtype="float32").unsqueeze(axis=0)
+            logger.info(
+                f"process the audio feature success, feat shape: {x_chunk.shape}"
+            )
+            return x_chunk, x_chunk_lens
 
 
 class ASREngine(BaseEngine):
@@ -310,7 +395,10 @@ class ASREngine(BaseEngine):
         logger.info("Initialize ASR server engine successfully.")
         return True
 
-    def preprocess(self, samples, sample_rate):
+    def preprocess(self,
+                   samples,
+                   sample_rate,
+                   model_type="deepspeech2online_aishell-zh-16k"):
         """preprocess
 
         Args:
@@ -321,6 +409,7 @@ class ASREngine(BaseEngine):
             x_chunk (numpy.array): shape[B, T, D]
             x_chunk_lens (numpy.array): shape[B]
         """
+        # if "deepspeech" in model_type:
         x_chunk, x_chunk_lens = self.executor.extract_feat(samples, sample_rate)
         return x_chunk, x_chunk_lens
 

paddlespeech/server/tests/asr/online/websocket_client.py

@@ -103,7 +103,7 @@ class ASRAudioHandler:
 def main(args):
     logging.basicConfig(level=logging.INFO)
     logging.info("asr websocket client start")
-    handler = ASRAudioHandler("127.0.0.1", 8090)
+    handler = ASRAudioHandler("127.0.0.1", 8096)
     loop = asyncio.get_event_loop()
 
     # support to process single audio file

paddlespeech/server/ws/asr_socket.py

@@ -14,6 +14,7 @@
 import json
 
 import numpy as np
+import json
 from fastapi import APIRouter
 from fastapi import WebSocket
 from fastapi import WebSocketDisconnect
@@ -28,7 +29,7 @@ router = APIRouter()
 
 @router.websocket('/ws/asr')
 async def websocket_endpoint(websocket: WebSocket):
-
+    print("websocket protocal receive the dataset")
     await websocket.accept()
 
     engine_pool = get_engine_pool()
@@ -36,14 +37,18 @@ async def websocket_endpoint(websocket: WebSocket):
     # init buffer
     chunk_buffer_conf = asr_engine.config.chunk_buffer_conf
     chunk_buffer = ChunkBuffer(
+        frame_duration_ms=chunk_buffer_conf['frame_duration_ms'],
         sample_rate=chunk_buffer_conf['sample_rate'],
         sample_width=chunk_buffer_conf['sample_width'])
     # init vad
-    vad_conf = asr_engine.config.vad_conf
-    vad = VADAudio(
-        aggressiveness=vad_conf['aggressiveness'],
-        rate=vad_conf['sample_rate'],
-        frame_duration_ms=vad_conf['frame_duration_ms'])
+    # print(asr_engine.config)
+    # print(type(asr_engine.config))
+    vad_conf = asr_engine.config.get('vad_conf', None)
+    if vad_conf:
+        vad = VADAudio(
+            aggressiveness=vad_conf['aggressiveness'],
+            rate=vad_conf['sample_rate'],
+            frame_duration_ms=vad_conf['frame_duration_ms'])
 
     try:
         while True:
@@ -65,7 +70,7 @@ async def websocket_endpoint(websocket: WebSocket):
                     engine_pool = get_engine_pool()
                     asr_engine = engine_pool['asr']
                     # reset single  engine for an new connection
-                    asr_engine.reset()
+                    # asr_engine.reset()
                     resp = {"status": "ok", "signal": "finished"}
                     await websocket.send_json(resp)
                     break
@@ -75,16 +80,16 @@ async def websocket_endpoint(websocket: WebSocket):
             elif "bytes" in message:
                 message = message["bytes"]
 
-                # vad for input bytes audio
-                vad.add_audio(message)
-                message = b''.join(f for f in vad.vad_collector()
-                                   if f is not None)
-
+                # # vad for input bytes audio
+                # vad.add_audio(message)
+                # message = b''.join(f for f in vad.vad_collector()
+                #                    if f is not None)
                 engine_pool = get_engine_pool()
                 asr_engine = engine_pool['asr']
                 asr_results = ""
                 frames = chunk_buffer.frame_generator(message)
                 for frame in frames:
+                    # get the pcm data from the bytes
                     samples = np.frombuffer(frame.bytes, dtype=np.int16)
                     sample_rate = asr_engine.config.sample_rate
                     x_chunk, x_chunk_lens = asr_engine.preprocess(samples,